Telecommunications

Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.

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I have a remote extension (yealink handset) that is dropping outbound calls after 35 seconds. I have diagnosed through logging that the asterisk server is not receiving the proper ACK, and the reason for that is that the asterisk server is looking for the reply from 192.168.1.4, which is the local subnet of the remote phone. Obviously the asterisk server will never receive a reply from that address. What changes do I need to make so that the asterisk server is looking for replies from the WAN IP of the remote network? NAT is setup properly on both ends. For reference, this is CompletePBX from Xorcom that we're dealing with and a copy of the asterisk log error is below.

[2018-08-16 09:27:08] WARNING[5720] chan_sip.c: Retransmission timeout reached on transmission 405286438@192.168.1.4 for seqno 2 (Critical Response)
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Cloud Class® Course: MCSA MCSE Windows Server 2012
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Cloud Class® Course: MCSA MCSE Windows Server 2012

This course teaches how to install and configure Windows Server 2012 R2.  It is the first step on your path to becoming a Microsoft Certified Solutions Expert (MCSE).

A customer of mine is moving their people from Office A to Office B, but is leaving the servers at Office A.

There are 5 networks in 3 different security zones that need to exist at Office B for the users, but the companies procurement department has vetoed using a Layer 2 type service because it's not in the current contract with their Telco provider.  :/

Net 1 - 100mb - Security Zone 1
Net 2 - 100mb - Security  Zone 1
Net 3 - 100mb -  Security Zone 2
Net 4 - 10mb -  Security Zone 3
Net 5 - 10mb - Security  Zone 3

The nets are all class C.  All the systems at Office B are new, and having different IP addresses is (amazingly / apparently) not a problem.

The Telco provider has sold them a pair of diverse 300 MB Ethernet MPLS circuits with three private VPNs which will be connected to two new Cisco 4431's at Office A and Office B that will both connect to all 5 network switches at Office A.  These routers will not directly route any packets between the 5 nets / 3 zones (there are other existing routers and firewalls at Office A that will do those tasks if needed), they are purely for connectivity between Office A and Office B.

Within the order notes for the circuits, I see the wording "Each MPLS CE router will utilize the Multi-VRF feature to segment traffic by application.  A total of 3 VPN's will be utilized."

I have no problems with routing protocol features (EIGRP or BGP) on high availability networks, and although I have never configured it …
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I am converting to SIP at a company site.  I set up the fortigate 60d identical to another one of our company sites where SIP is already implemented.

The SIP Cloud provider tells me that the firewall is not "passing confirmation packets  the phone system (PBX) sends out when it detects an incoming call",
and this is causing their system to transfer the call to our failover number.

I have opened up all requested ports (TCP/UDP etc), configurd policy,and QOS to high priority- everything they suggested, and as I mentioned earlier, it is configured exactly like our other site's fw.

I suspect it is a configuration with one of their servers.  In any case, Logging for all events is enabled in the firewall policy.  How can I tell if the firewall is blocking/not passing back the confirmation packets they SIp provider mentions?
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We have been using Skype for Business (formerly Lync) for IM and video conferencing but are thinking about using Skype PSTN as a replacement for our current PBX.

We are in the process of organising a trial for some users to pilot Skype for Business as their only phone option.

One of the things we need to do is gather some data once their trial has finished and I am looking for help identifying test criteria for the new system.

If you have any thoughts about the sort of questions we should be asking users to assess the suitability of Skype for Business, then please let me know.

Thoughts so far:-

Equipment
Which microphone device do you normally use for S4B phone calls?
Which amplification device do you normally use for S4B phone calls?
Frequency of use
In a typical day, how likely are you to make a phonecall using S4B
In a typical day, how likely are you to receive a phonecall using S4B
Sound Quality
How does the sound quality compare to landline
How does it compare to mobile

Any questions / comments appreciated.


Jon
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When I call my cell phone using skype on my windows 10 computer
I see the caller id that I selected on skype.com
I am spoofing my phone number to display another phone number that I use but am not currently using because I am calling from a computer.

But when I call a big business the number that shows up is not mine and can not be called again.

How do I purchase the caller id system that a big business has.
Caller ID that does not work with a spoofed phone number.
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good day,

I have two switch both configured for vlan 869 (Voice). The phones connected to SW01 can get IP address and phones connected to SW02 cannot get IP address. Can someone assist where I made configuration mistake. This is my first config as I am learning the Cisco commands. Attached are my configurations for both switch.
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Hi, we have a data network here with Cisco switches that we manage. Now, there is a also a VOIP vendor who his own switches in our network. (He didnt want to use ours and VLAN everything).

Both the networks are on different subnets. Now, we noticed all of a sudden PC's started getting IP's from the phone subnet...and it's wreaking havoc internally. I tried to manually trace all 200+ cables in the office to see if someone plugged a phone device into the data network, but no luck..

How else can i troubleshoot this from say a switch level?
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For people that own / manage 800 numbers for businesses:

Do you get charged for / pay for calls from payphones?  Years ago there was a charge - 26c I think it was - that people that own toll free numbers were charged that went to the pay phone owner.  Wonder if that's still in effect. And for bigger businesses, do they (the 800 number supplier) not charge the 800 number owner?

And 800 numbers - does the owner typically pay a per minute cost? Or has it moved like outgoing phone lines to a flat fee / month?

thanks!
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help with setting up VLAN on a few switches for phone and data equipment.

i have a series of managed switches that are uplinked together.  I would like to set up a VLAN 100 for a dedicated router that is on port 48 of one switch. This router will listen to requests from phones that are plugged into any other random ports on the switch.  This switch is a ubiquiti unit that allows me to set port 48 to listen to vlan 100 traffic only.

The phones are set to 802.1Q with a vlan of 100.  there are other computers and servers on the switch that are on a 192.168.0.x subnet.  The server is handing out DHCP as well as the router on port 48.  The idea is to isolate the traffic for the phones to ONLY communicate with the DHCP server on port 48.  

Right now, this setting is working. However my question to you, is since the phones are all plugged into random ports 1-47 and set to vlan100  and these ports are set to listen to both default lan traffic as well as vlan100...am i simply congesting the switch with added default and vlan traffic vs setting the actual ports that the phones are plugged into to ONLY vlan 100?

Also, if i plug in another switch,, do i need to set the uplink from one switch to another switch with a vlan100 for them to comminicate or will they pass the phones traffic that is tagged 802.1Q VLAN 100 traffic to the other where the port 48 will ultimately listen and grab it? Thank you!
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Is there a converter or way to utilize a very nice slightly used Panasonic phone system that does not have VOIP technology built in to the phone system. It's perfect for a small business. Bought it a few years ago as the VOIPs were coming in. Wish I had not now. Unfortunately I have a cell phone but would like to utilize the home phone system NOT VOIP again because its conveniently setup to two different work stations. So I was wondering if the phone system I have can still be utilized with a phone line though some kind of a converter or am I forced to purchase all new phone system that works with VOIP? Because I am under contract with AT&T cellular. I unfortunately cannot switch and AT&T does not have landline services in my area so that means I have to get all new services with another provider? It just mean redundancy I just do not want to put everything through my cell phone. when I have a super nice phone system in house. Your thoughts appreciated.
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Cloud Class® Course: Amazon Web Services - Basic
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Cloud Class® Course: Amazon Web Services - Basic

Are you thinking about creating an Amazon Web Services account for your business? Not sure where to start? In this course you’ll get an overview of the history of AWS and take a tour of their user interface.

Cisco IP Phone 7941 still trying to upgrade.

Physically took phone to TFTP Server and uploaded current OS software to the phone.

Everthing in Call Manager looks good.

Cleared port security on the switch.

Phone daisy chained to PC.

PC has good Internet/Network connectivity.
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I rewired my house with CAT6, and have a problem terminating the jacks at the end.

I bought IDEAL CAT6 jacks, which i've never really used before.  

I spent several hours on the CAT6 jacks, and I found it to be very uncomfortable.  the inner core is a nuisance -- i know it's for the better(less attenuation?)
the wires fit into the jack very loosely and i could barely guide the 8 wires in order into the terminating jack.  Once I crimped it, it was fairly loose as well -- i could pull and without exerting so much effort pull the wire back out.

i wired it once, tested it -- it failed -- no signal whatsoever.
i didn't know what was wrong and re-wired it a seocnd time, and still uncomfortable -- and failed again.

i searched for documentation and the internet is innuntdated with info on the old system using CAT5/CAT5e, and many videos even claim to show how to terminate CAT6, but don't get into the sled and liner.

i tried once more with the sled and liner using a poor IDEAL schematic, and then gave up.

after a while, of frustration i thought why not just try the CAT5e jacks -- and in 5 minutes I did what took me 4 hours previously.  felt comfortable --slid in snug, crimped strong. and tested perfectly.

my question:

can anyone provide a detailed CAT6 patch termination document including whether or not the sled and liner are necessary, and any other specifactions ( i think the wires didn't even fit in the liner)

my second question:

is performance going…
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Hello,
I have Avaya ip office in branch a and Nortel (upgraded to avaya) in branch b
I was connect branch b  to branch a with billion ssl
no its working with mikrotik sstp connection.
avaya can ping to Nortel and Nortel can ping to Avaya but the sip is not working between them
how I can resolve that ?
is there any documentation about that ?
thanks.
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I am looking for a Program or Script to dial remote Hayes AT Modems and log results. We have a large number of Hayes compatible modems connected to Verizon POTS lines. Many of these modems are in locations where there are frequent service interruptions. We would like to automate testing the availability and logging results. Is anyone aware of how we can do this?
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Look for outsource call center that they will have to use my five9 subscription to do in and out bounds calls.
https://www.five9.com/

Do u know any places that can provide those services?
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Hi,
I have a Panasonic TDE phone systeme and one of my phones, a KX-DT343, cannot do External Call. It says "Restreint".  
I am in the PBX software and cannot find where a restriction could have been implemented.
In a https://panasonic.ca/pcs/operatinginstructions/kxts620-oi-fr.pdf phone manual I found that the Restrict fonction is activated ON the phone itself but on my kx-dt343s there is no option visible for this.
It can be accessible from the phone or from the PBX  I guess but where?
Tx!
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I am running FreePBX v13 and need to hangup all current calls from a set of extensions at a set time every day.  
Does anyone have a script or method that could perform this type of function??
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Would it be a lot of data loss from SM to MM?
We have fiber from ISP [10Gb] is 9/125 SM going from SC to LC then into our LC 50/125 MM then into SM switch network module with SFP.

Would I have a lot of issues with that setup?

 I also read some people are using mode conditioning cables.  Will that help a lot?
https://community.fs.com/blog/mode-conditioning-patch-cord-utilized-in-gigabit-ethernet-applications.html
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SIP Provider Review, I have quote from Access Point Inc, I've never hear from the company before and bit skeptical where all their servers are. I don't to want SIP server siting on West end of country when our building is at East End.  Is anyone who experience with them? any feed back? or SIP provider who does good what they do?
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Cloud Class® Course: SQL Server Core 2016
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Cloud Class® Course: SQL Server Core 2016

This course will introduce you to SQL Server Core 2016, as well as teach you about SSMS, data tools, installation, server configuration, using Management Studio, and writing and executing queries.

How can you tell what DIDs are associated with an AT&T PRI? I've migrated pretty much all of my traffic to a SIP circuit and I want to start cancelling circuits.
Should the circuit ID to DID mapping be on the bills?
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I have a BCM 200 with 4 analog lines incoming.  I would like to use the BCM to call forward these 4 lines into a new phone system that I am trialing. Could someone please point me in the right direction as I am not familiar with configuring BCM's.
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Skype (Windows 10). The Video Button has disappeared from its position on right side top - alongside the telephone icon. Have tried for answers without success. What does this signify?
Has it anything to do with paying money.  It is not a problem with my computer, as I have opened the account on a second computer with same result.
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Hello,

I am looking for an entreprise grade gsm modem compatible with kannel.
I need to send at  least 100 sms per minute.
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Dear Experts, can you suggest the pros and cons of Skype for Business? Does it have these features in Broadcast mode?
- Share file
- Share screen
- Control screen

Compared to other opponent like ClickMeeting, Zoom, Webex; what are its advantages and disadvantages? Many thanks!
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I always get confused / use firewall / router rules so infrequently that I don't know  what the right way to set these things up.

Can you help?

I have a VOIP service using a Grandstream HT-502 V1.2A gadget.  The call quality isn't always good.  That device is plugged directly into one of the ports on an actiontec router (left over from when we had verizon fios, but now we have cable internet 15down / 5 up speed.

I have lots of other devices plugged into a gig network switch in the basement that might be using the internet at the same time as the calls?  That gig network switch has 1 cable going over to the actiontec also (so there's only 2 cables on the lan side of the router).

To improve call quality, that's a job for QoS, right?

The attached picture is what I did in the actiontec router.  The grandstream has the ip of 192.168.1.52

But then i thought, should this be on the Ethernet/Coax or  Broadband Connection (Ethernet/Coax) sections? Did I at least get outbound rather than inbound correct?

But that just gets the call out of the house with highest priority.  Once it's on the web, it's fighting with all kinds of data / can't prioritize it, right?

Does the VOIP provider have any bearing on the quality of the calls? Iwe are using VOIPO.com).  is there  a way to substantiate / test where the poor qiuality - dropped fractions of a second in the conversation, etc.  NO stuttering, max headroom type things.  Just a m ssing sound here and there.

And most …
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Telecommunications

Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.